A family of digital subscriber line technologies (xDSL) provide high-speed digital data transmission over telephone lines. Asymmetric digital subscriber line (ADSL) is one type of DSL technology. ADSL is widely used to carry information (such as voice and/or data) in a variety of residential and business customer applications.
ADSL modems are often provided at customer premises and a local exchange (such as a central office or private branch exchange). The ADSL modems are coupled to one another over a local loop. In a common implementation, an ADSL local loop is made up of a pair of wires and can simultaneously transport information on three channels: (1) a high-speed downstream digital channel towards a customer, (2) a medium speed upstream digital channel towards a local exchange, and (3) a plain old telephone service (POTS) or Integrated Services Digital Network (ISDN) channel. Because the bit rate on the downstream channel is often higher than the bit rate on the upstream channel, ADSL is termed “asymmetric.”
Different standards have been used to define versions of ADSL technology. Early standards included ANSI T1.413 and ITU G.992.1. ANSI T1.413 and ITU G.992.1, among other things, require splitters at ends of a DSL local loop to separate the POTS channel and digital ADSL channels. Another version of ADSL is ADSL Lite or Universal ADSL defined by standard ITU G.992.2 or simply “G-Lite” which does not require splitters. Newer versions called ADSL2 operate at even higher bits rates and are defined by a standard ITU G.992.3 (or G.DMT.bis) which includes splitters, and standard ITU G.992.4 (or G.Lite.bis) which does not include splitters.
In certain ADSL standards, such as T1.413 issue 2 and ITU-T G.992.1/2/3/4, a pseudo random signal called Medley is used to measure the Signal-To-Noise Ratio (SNR) at the receiver on the various tones of multi-carrier systems. According to this SNR, a bit loading is done to determine the line capacity. This Medley signal can also be used to train adaptive algorithms at the receiver (e.g. equalizer) or at the transmitter (e.g. echo cancellers tap).
In all of these conventional standards, the Medley signal is generated using a pseudo-random bit sequence (PRBS), determined by a fixed polynomial and initial state that will modulate four-quadrature amplitude modulated (4QAM) symbols on each tone before modulation by an inverse discrete fast Fourier transform (IFFT). For example, the G.992.3 standard (incorporated by reference herein in its entirety) defines sequences, which are known as C-MEDLEY and R-MEDLEY. Appendix A attached hereto and incorporated by reference in its entirety herein illustrates examples of this fixed parameter approach in more detail.
Unfortunately, conventional bit sequences, for a given set of used tones or maximum number of upstream and downstream carriers (i.e., NSCus, NSCds), can produce (after IFFT processing) a high Peak-to-Average Ratio (PAR), which is the ratio between the maximum amplitude of the signal and the root-mean-square (RMS) amplitude. A high PAR can potentially lead to clipping of the signal in the digital or analog domain. This clipping usually impairs the estimation of the SNR per bin at the receiver, and the adaptation of algorithms at the receiver or transmitter. The PAR of the analog signal can be rather different from the PAR at the IFFT output because of digital and analog filtering.
What is needed are methods and systems for improved bit sequence generation in multi-carrier communication systems.